QUIC J. Iyengar, Ed.
Internet-Draft I. Swett, Ed.
Intended status: Standards Track Google
Expires: August 1, 2018 January 28, 2018
QUIC Loss Detection and Congestion Controldraft-ietf-quic-recovery-09
Abstract
This document describes loss detection and congestion control
mechanisms for QUIC.
Note to Readers
Discussion of this draft takes place on the QUIC working group
mailing list (quic@ietf.org), which is archived at
https://mailarchive.ietf.org/arch/search/?email_list=quic [1].
Working Group information can be found at https://github.com/quicwg
[2]; source code and issues list for this draft can be found at
https://github.com/quicwg/base-drafts/labels/-recovery [3].
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on August 1, 2018.
Copyright Notice
Copyright (c) 2018 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
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Internet-Draft QUIC Loss Detection January 20182. Design of the QUIC Transmission Machinery
All transmissions in QUIC are sent with a packet-level header, which
includes a packet sequence number (referred to below as a packet
number). These packet numbers never repeat in the lifetime of a
connection, and are monotonically increasing, which prevents
ambiguity. This fundamental design decision obviates the need for
disambiguating between transmissions and retransmissions and
eliminates significant complexity from QUIC's interpretation of TCP
loss detection mechanisms.
Every packet may contain several frames. We outline the frames that
are important to the loss detection and congestion control machinery
below.
o Retransmittable frames are frames requiring reliable delivery.
The most common are STREAM frames, which typically contain
application data.
o Crypto handshake data is sent on stream 0, and uses the
reliability machinery of QUIC underneath.
o ACK frames contain acknowledgment information. ACK frames contain
one or more ranges of acknowledged packets.
2.1. Relevant Differences Between QUIC and TCP
Readers familiar with TCP's loss detection and congestion control
will find algorithms here that parallel well-known TCP ones.
Protocol differences between QUIC and TCP however contribute to
algorithmic differences. We briefly describe these protocol
differences below.
2.1.1. Monotonically Increasing Packet Numbers
TCP conflates transmission sequence number at the sender with
delivery sequence number at the receiver, which results in
retransmissions of the same data carrying the same sequence number,
and consequently to problems caused by "retransmission ambiguity".
QUIC separates the two: QUIC uses a packet number for transmissions,
and any data that is to be delivered to the receiving application(s)
is sent in one or more streams, with delivery order determined by
stream offsets encoded within STREAM frames.
QUIC's packet number is strictly increasing, and directly encodes
transmission order. A higher QUIC packet number signifies that the
packet was sent later, and a lower QUIC packet number signifies that
the packet was sent earlier. When a packet containing frames is
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deemed lost, QUIC rebundles necessary frames in a new packet with a
new packet number, removing ambiguity about which packet is
acknowledged when an ACK is received. Consequently, more accurate
RTT measurements can be made, spurious retransmissions are trivially
detected, and mechanisms such as Fast Retransmit can be applied
universally, based only on packet number.
This design point significantly simplifies loss detection mechanisms
for QUIC. Most TCP mechanisms implicitly attempt to infer
transmission ordering based on TCP sequence numbers - a non-trivial
task, especially when TCP timestamps are not available.
2.1.2. No Reneging
QUIC ACKs contain information that is similar to TCP SACK, but QUIC
does not allow any acked packet to be reneged, greatly simplifying
implementations on both sides and reducing memory pressure on the
sender.
2.1.3. More ACK Ranges
QUIC supports many ACK ranges, opposed to TCP's 3 SACK ranges. In
high loss environments, this speeds recovery, reduces spurious
retransmits, and ensures forward progress without relying on
timeouts.
2.1.4. Explicit Correction For Delayed Acks
QUIC ACKs explicitly encode the delay incurred at the receiver
between when a packet is received and when the corresponding ACK is
sent. This allows the receiver of the ACK to adjust for receiver
delays, specifically the delayed ack timer, when estimating the path
RTT. This mechanism also allows a receiver to measure and report the
delay from when a packet was received by the OS kernel, which is
useful in receivers which may incur delays such as context-switch
latency before a userspace QUIC receiver processes a received packet.
3. Loss Detection
QUIC senders use both ack information and timeouts to detect lost
packets, and this section provides a description of these algorithms.
Estimating the network round-trip time (RTT) is critical to these
algorithms and is described first.
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Internet-Draft QUIC Loss Detection January 20183.1. Computing the RTT estimate
RTT is calculated when an ACK frame arrives by computing the
difference between the current time and the time the largest newly
acked packet was sent. If no packets are newly acknowledged, RTT
cannot be calculated. When RTT is calculated, the ack delay field
from the ACK frame SHOULD be subtracted from the RTT as long as the
result is larger than the Min RTT. If the result is smaller than the
min_rtt, the RTT should be used, but the ack delay field should be
ignored.
Like TCP, QUIC calculates both smoothed RTT and RTT variance as
specified in [RFC6298].
Min RTT is the minimum RTT measured over the connection, prior to
adjusting by ack delay. Ignoring ack delay for min RTT prevents
intentional or unintentional underestimation of min RTT, which in
turn prevents underestimating smoothed RTT.
3.2. Ack-based Detection
Ack-based loss detection implements the spirit of TCP's Fast
Retransmit [RFC5681], Early Retransmit [RFC5827], FACK, and SACK loss
recovery [RFC6675]. This section provides an overview of how these
algorithms are implemented in QUIC.
(TODO: Define unacknowledged packet, ackable packet, outstanding
bytes.)
3.2.1. Fast Retransmit
An unacknowledged packet is marked as lost when an acknowledgment is
received for a packet that was sent a threshold number of packets
(kReorderingThreshold) after the unacknowledged packet. Receipt of
the ack indicates that a later packet was received, while
kReorderingThreshold provides some tolerance for reordering of
packets in the network.
The RECOMMENDED initial value for kReorderingThreshold is 3.
We derive this default from recommendations for TCP loss recovery
[RFC5681] [RFC6675]. It is possible for networks to exhibit higher
degrees of reordering, causing a sender to detect spurious losses.
Detecting spurious losses leads to unnecessary retransmissions and
may result in degraded performance due to the actions of the
congestion controller upon detecting loss. Implementers MAY use
algorithms developed for TCP, such as TCP-NCR [RFC4653], to improve
QUIC's reordering resilience, though care should be taken to map TCP
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specifics to QUIC correctly. Similarly, using time-based loss
detection to deal with reordering, such as in PR-TCP, should be more
readily usable in QUIC. Making QUIC deal with such networks is
important open research, and implementers are encouraged to explore
this space.
3.2.2. Early Retransmit
Unacknowledged packets close to the tail may have fewer than
kReorderingThreshold number of ackable packets sent after them. Loss
of such packets cannot be detected via Fast Retransmit. To enable
ack-based loss detection of such packets, receipt of an
acknowledgment for the last outstanding ackable packet triggers the
Early Retransmit process, as follows.
If there are unacknowledged ackable packets still pending, they ought
to be marked as lost. To compensate for the reduced reordering
resilience, the sender SHOULD set an alarm for a small period of
time. If the unacknowledged ackable packets are not acknowledged
during this time, then these packets MUST be marked as lost.
An endpoint SHOULD set the alarm such that a packet is marked as lost
no earlier than 1.25 * max(SRTT, latest_RTT) since when it was sent.
Using max(SRTT, latest_RTT) protects from the two following cases:
o the latest RTT sample is lower than the SRTT, perhaps due to
reordering where packet whose ack triggered the Early Retransit
process encountered a shorter path;
o the latest RTT sample is higher than the SRTT, perhaps due to a
sustained increase in the actual RTT, but the smoothed SRTT has
not yet caught up.
The 1.25 multiplier increases reordering resilience. Implementers
MAY experiment with using other multipliers, bearing in mind that a
lower multiplier reduces reordering resilience and increases spurious
retransmissions, and a higher multipler increases loss recovery
delay.
This mechanism is based on Early Retransmit for TCP [RFC5827].
However, [RFC5827] does not include the alarm described above. Early
Retransmit is prone to spurious retransmissions due to its reduced
reordering resilence without the alarm. This observation led Linux
TCP implementers to implement an alarm for TCP as well, and this
document incorporates this advancement.
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Internet-Draft QUIC Loss Detection January 20183.3. Timer-based Detection
Timer-based loss detection implements the spirit of TCP's Tail Loss
Probe and Retransmission Timeout mechanisms.
3.3.1. Tail Loss Probe
The algorithm described in this section is an adaptation of the Tail
Loss Probe algorithm proposed for TCP [TLP].
A packet sent at the tail is particularly vulnerable to slow loss
detection, since acks of subsequent packets are needed to trigger
ack-based detection. To ameliorate this weakness of tail packets,
the sender schedules an alarm when the last ackable packet before
quiescence is transmitted. When this alarm fires, a Tail Loss Probe
(TLP) packet is sent to evoke an acknowledgement from the receiver.
The alarm duration, or Probe Timeout (PTO), is set based on the
following conditions:
o PTO SHOULD be scheduled for max(1.5*SRTT+MaxAckDelay, 10ms)
o If RTO (Section 3.3.2) is earlier, schedule a TLP alarm in its
place. That is, PTO SHOULD be scheduled for min(RTO, PTO).
MaxAckDelay is the maximum ack delay supplied in an incoming ACK
frame. MaxAckDelay excludes ack delays that aren't included in an
RTT sample because they're too large and excludes those which
reference an ack-only packet.
QUIC diverges from TCP by calculating MaxAckDelay dynamically,
instead of assuming a constant delayed ack timeout for all
connections. QUIC includes this in all probe timeouts, because it
assume the ack delay may come into play, regardless of the number of
packets outstanding. TCP's TLP assumes if at least 2 packets are
outstanding, acks will not be delayed.
A PTO value of at least 1.5*SRTT ensures that the ACK is overdue.
The 1.5 is based on [LOSS-PROBE], but implementations MAY experiment
with other constants.
To reduce latency, it is RECOMMENDED that the sender set and allow
the TLP alarm to fire twice before setting an RTO alarm. In other
words, when the TLP alarm fires the first time, a TLP packet is sent,
and it is RECOMMENDED that the TLP alarm be scheduled for a second
time. When the TLP alarm fires the second time, a second TLP packet
is sent, and an RTO alarm SHOULD be scheduled Section 3.3.2.
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A TLP packet SHOULD carry new data when possible. If new data is
unavailable or new data cannot be sent due to flow control, a TLP
packet MAY retransmit unacknowledged data to potentially reduce
recovery time. Since a TLP alarm is used to send a probe into the
network prior to establishing any packet loss, prior unacknowledged
packets SHOULD NOT be marked as lost when a TLP alarm fires.
A TLP packet MUST NOT be blocked by the sender's congestion
controller. The sender MUST however count these bytes as additional
bytes in flight, since a TLP adds network load without establishing
packet loss.
A sender may not know that a packet being sent is a tail packet.
Consequently, a sender may have to arm or adjust the TLP alarm on
every sent ackable packet.
3.3.2. Retransmission Timeout
A Retransmission Timeout (RTO) alarm is the final backstop for loss
detection. The algorithm used in QUIC is based on the RTO algorithm
for TCP [RFC5681] and is additionally resilient to spurious RTO
events [RFC5682].
When the last TLP packet is sent, an alarm is scheduled for the RTO
period. When this alarm fires, the sender sends two packets, to
evoke acknowledgements from the receiver, and restarts the RTO alarm.
Similar to TCP [RFC6298], the RTO period is set based on the
following conditions:
o When the final TLP packet is sent, the RTO period is set to
max(SRTT + 4*RTTVAR + MaxAckDelay, minRTO)
o When an RTO alarm fires, the RTO period is doubled.
The sender typically has incurred a high latency penalty by the time
an RTO alarm fires, and this penalty increases exponentially in
subsequent consecutive RTO events. Sending a single packet on an RTO
event therefore makes the connection very sensitive to single packet
loss. Sending two packets instead of one significantly increases
resilience to packet drop in both directions, thus reducing the
probability of consecutive RTO events.
QUIC's RTO algorithm differs from TCP in that the firing of an RTO
alarm is not considered a strong enough signal of packet loss, so
does not result in an immediate change to congestion window or
recovery state. An RTO alarm fires only when there's a prolonged
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period of network silence, which could be caused by a change in the
underlying network RTT.
QUIC also diverges from TCP by including MaxAckDelay in the RTO
period. QUIC is able to explicitly model delay at the receiver via
the ack delay field in the ACK frame. Since QUIC corrects for this
delay in its SRTT and RTTVAR computations, it is necessary to add
this delay explicitly in the TLP and RTO computation.
When an acknowledgment is received for a packet sent on an RTO event,
any unacknowledged packets with lower packet numbers than those
acknowledged MUST be marked as lost.
A packet sent when an RTO alarm fires MAY carry new data if available
or unacknowledged data to potentially reduce recovery time. Since
this packet is sent as a probe into the network prior to establishing
any packet loss, prior unacknowledged packets SHOULD NOT be marked as
lost.
A packet sent on an RTO alarm MUST NOT be blocked by the sender's
congestion controller. A sender MUST however count these bytes as
additional bytes in flight, since this packet adds network load
without establishing packet loss.
3.3.3. Handshake Timeout
Handshake packets, which contain STREAM frames for stream 0, are
critical to QUIC transport and crypto negotiation, so a separate
alarm is used for them.
The initial handshake timeout SHOULD be set to twice the initial RTT.
At the beginning, there are no prior RTT samples within a connection.
Resumed connections over the same network SHOULD use the previous
connection's final smoothed RTT value as the resumed connection's
initial RTT.
If no previous RTT is available, or if the network changes, the
initial RTT SHOULD be set to 100ms.
When the first handshake packet is sent, the sender SHOULD set an
alarm for the handshake timeout period.
When the alarm fires, the sender MUST retransmit all unacknowledged
handshake data. On each consecutive firing of the handshake alarm,
the sender SHOULD double the handshake timeout and set an alarm for
this period.
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When an acknowledgement is received for a handshake packet, the new
RTT is computed and the alarm SHOULD be set for twice the newly
computed smoothed RTT.
Handshake data may be cancelled by handshake state transitions. In
particular, all non-protected data SHOULD no longer be transmitted
once packet protection is available.
(TODO: Work this section some more. Add text on client vs. server,
and on stateless retry.)
3.4. Pseudocode3.4.1. Constants of interest
Constants used in loss recovery are based on a combination of RFCs,
papers, and common practice. Some may need to be changed or
negotiated in order to better suit a variety of environments.
kMaxTLPs (default 2): Maximum number of tail loss probes before an
RTO fires.
kReorderingThreshold (default 3): Maximum reordering in packet
number space before FACK style loss detection considers a packet
lost.
kTimeReorderingFraction (default 1/8): Maximum reordering in time
space before time based loss detection considers a packet lost.
In fraction of an RTT.
kUsingTimeLossDetection (default false): Whether time based loss
detection is in use. If false, uses FACK style loss detection.
kMinTLPTimeout (default 10ms): Minimum time in the future a tail
loss probe alarm may be set for.
kMinRTOTimeout (default 200ms): Minimum time in the future an RTO
alarm may be set for.
kDelayedAckTimeout (default 25ms): The length of the peer's delayed
ack timer.
kDefaultInitialRtt (default 100ms): The default RTT used before an
RTT sample is taken.
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Internet-Draft QUIC Loss Detection January 20183.4.2. Variables of interest
Variables required to implement the congestion control mechanisms are
described in this section.
loss_detection_alarm: Multi-modal alarm used for loss detection.
handshake_count: The number of times the handshake packets have been
retransmitted without receiving an ack.
tlp_count: The number of times a tail loss probe has been sent
without receiving an ack.
rto_count: The number of times an rto has been sent without
receiving an ack.
largest_sent_before_rto: The last packet number sent prior to the
first retransmission timeout.
time_of_last_sent_packet: The time the most recent packet was sent.
largest_sent_packet: The packet number of the most recently sent
packet.
largest_acked_packet: The largest packet number acknowledged in an
ACK frame.
latest_rtt: The most recent RTT measurement made when receiving an
ack for a previously unacked packet.
smoothed_rtt: The smoothed RTT of the connection, computed as
described in [RFC6298]
rttvar: The RTT variance, computed as described in [RFC6298]
min_rtt: The minimum RTT seen in the connection, ignoring ack delay.
max_ack_delay: The maximum ack delay in an incoming ACK frame for
this connection. Excludes ack delays for ack only packets and
those that create an RTT sample less than min_rtt.
reordering_threshold: The largest delta between the largest acked
retransmittable packet and a packet containing retransmittable
frames before it's declared lost.
time_reordering_fraction: The reordering window as a fraction of
max(smoothed_rtt, latest_rtt).
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loss_time: The time at which the next packet will be considered lost
based on early transmit or exceeding the reordering window in
time.
sent_packets: An association of packet numbers to information about
them, including a number field indicating the packet number, a
time field indicating the time a packet was sent, a boolean
indicating whether the packet is ack only, and a bytes field
indicating the packet's size. sent_packets is ordered by packet
number, and packets remain in sent_packets until acknowledged or
lost.
3.4.3. Initialization
At the beginning of the connection, initialize the loss detection
variables as follows:
loss_detection_alarm.reset()
handshake_count = 0
tlp_count = 0
rto_count = 0
if (kUsingTimeLossDetection)
reordering_threshold = infinite
time_reordering_fraction = kTimeReorderingFraction
else:
reordering_threshold = kReorderingThreshold
time_reordering_fraction = infinite
loss_time = 0
smoothed_rtt = 0
rttvar = 0
min_rtt = 0
max_ack_delay = 0
largest_sent_before_rto = 0
time_of_last_sent_packet = 0
largest_sent_packet = 0
3.4.4. On Sending a Packet
After any packet is sent, be it a new transmission or a rebundled
transmission, the following OnPacketSent function is called. The
parameters to OnPacketSent are as follows:
o packet_number: The packet number of the sent packet.
o is_ack_only: A boolean that indicates whether a packet only
contains an ACK frame. If true, it is still expected an ack will
be received for this packet, but it is not congestion controlled.
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o sent_bytes: The number of bytes sent in the packet, not including
UDP or IP overhead, but including QUIC framing overhead.
Pseudocode for OnPacketSent follows:
OnPacketSent(packet_number, is_ack_only, sent_bytes):
time_of_last_sent_packet = now
largest_sent_packet = packet_number
sent_packets[packet_number].packet_number = packet_number
sent_packets[packet_number].time = now
sent_packets[packet_number].ack_only = is_ack_only
if !is_ack_only:
OnPacketSentCC(sent_bytes)
sent_packets[packet_number].bytes = sent_bytes
SetLossDetectionAlarm()
3.4.5. On Ack Receipt
When an ack is received, it may acknowledge 0 or more packets.
Pseudocode for OnAckReceived and UpdateRtt follow:
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OnAckReceived(ack):
largest_acked_packet = ack.largest_acked
// If the largest acked is newly acked, update the RTT.
if (sent_packets[ack.largest_acked]):
latest_rtt = now - sent_packets[ack.largest_acked].time
UpdateRtt(latest_rtt, ack.ack_delay)
// Find all newly acked packets.
for acked_packet in DetermineNewlyAckedPackets():
OnPacketAcked(acked_packet.packet_number)
DetectLostPackets(ack.largest_acked_packet)
SetLossDetectionAlarm()
UpdateRtt(latest_rtt, ack_delay):
// min_rtt ignores ack delay.
min_rtt = min(min_rtt, latest_rtt)
// Adjust for ack delay if it's plausible.
if (latest_rtt - min_rtt > ack_delay):
latest_rtt -= ack_delay
// Only save into max ack delay if it's used
// for rtt calculation and is not ack only.
if (!sent_packets[ack.largest_acked].ack_only)
max_ack_delay = max(max_ack_delay, ack_delay)
// Based on {{RFC6298}}.
if (smoothed_rtt == 0):
smoothed_rtt = latest_rtt
rttvar = latest_rtt / 2
else:
rttvar_sample = abs(smoothed_rtt - latest_rtt)
rttvar = 3/4 * rttvar + 1/4 * rttvar_sample
smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * latest_rtt
3.4.6. On Packet Acknowledgment
When a packet is acked for the first time, the following
OnPacketAcked function is called. Note that a single ACK frame may
newly acknowledge several packets. OnPacketAcked must be called once
for each of these newly acked packets.
OnPacketAcked takes one parameter, acked_packet_number, which is the
packet number of the newly acked packet, and returns a list of packet
numbers that are detected as lost.
If this is the first acknowledgement following RTO, check if the
smallest newly acknowledged packet is one sent by the RTO, and if so,
inform congestion control of a verified RTO, similar to F-RTO
[RFC5682]
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Pseudocode for OnPacketAcked follows:
OnPacketAcked(acked_packet_number):
OnPacketAckedCC(acked_packet_number)
// If a packet sent prior to RTO was acked, then the RTO
// was spurious. Otherwise, inform congestion control.
if (rto_count > 0 &&
acked_packet_number > largest_sent_before_rto)
OnRetransmissionTimeoutVerified()
handshake_count = 0
tlp_count = 0
rto_count = 0
sent_packets.remove(acked_packet_number)
3.4.7. Setting the Loss Detection Alarm
QUIC loss detection uses a single alarm for all timer-based loss
detection. The duration of the alarm is based on the alarm's mode,
which is set in the packet and timer events further below. The
function SetLossDetectionAlarm defined below shows how the single
timer is set based on the alarm mode.
3.4.7.1. Handshake Alarm
When a connection has unacknowledged handshake data, the handshake
alarm is set and when it expires, all unacknowledgedd handshake data
is retransmitted.
When stateless rejects are in use, the connection is considered
immediately closed once a reject is sent, so no timer is set to
retransmit the reject.
Version negotiation packets are always stateless, and MUST be sent
once per handshake packet that uses an unsupported QUIC version, and
MAY be sent in response to 0RTT packets.
3.4.7.2. Tail Loss Probe and Retransmission Alarm
Tail loss probes [LOSS-PROBE] and retransmission timeouts [RFC6298]
are an alarm based mechanism to recover from cases when there are
outstanding retransmittable packets, but an acknowledgement has not
been received in a timely manner.
The TLP and RTO timers are armed when there is not unacknowledged
handshake data. The TLP alarm is set until the max number of TLP
packets have been sent, and then the RTO timer is set.
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OnLossDetectionAlarm():
if (handshake packets are outstanding):
// Handshake retransmission alarm.
RetransmitAllHandshakePackets()
handshake_count++
else if (loss_time != 0):
// Early retransmit or Time Loss Detection
DetectLostPackets(largest_acked_packet)
else if (tlp_count < kMaxTLPs):
// Tail Loss Probe.
SendOnePacket()
tlp_count++
else:
// RTO.
if (rto_count == 0)
largest_sent_before_rto = largest_sent_packet
SendTwoPackets()
rto_count++
SetLossDetectionAlarm()
3.4.9. Detecting Lost Packets
Packets in QUIC are only considered lost once a larger packet number
is acknowledged. DetectLostPackets is called every time an ack is
received. If the loss detection alarm fires and the loss_time is
set, the previous largest acked packet is supplied.
3.4.9.1. Handshake Packets
The receiver MUST close the connection with an error of type
OPTIMISTIC_ACK when receiving an unprotected packet that acks
protected packets. The receiver MUST trust protected acks for
unprotected packets, however. Aside from this, loss detection for
handshake packets when an ack is processed is identical to other
packets.
3.4.9.2. Pseudocode
DetectLostPackets takes one parameter, acked, which is the largest
acked packet.
Pseudocode for DetectLostPackets follows:
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congestion control is specified in bytes due to finer control and the
ease of appropriate byte counting [RFC3465].
4.1. Slow Start
QUIC begins every connection in slow start and exits slow start upon
loss. QUIC re-enters slow start anytime the congestion window is
less than sshthresh, which typically only occurs after an RTO. While
in slow start, QUIC increases the congestion window by the number of
acknowledged bytes when each ack is processed.
4.2. Congestion Avoidance
Slow start exits to congestion avoidance. Congestion avoidance in
NewReno uses an additive increase multiplicative decrease (AIMD)
approach that increases the congestion window by one MSS of bytes per
congestion window acknowledged. When a loss is detected, NewReno
halves the congestion window and sets the slow start threshold to the
new congestion window.
4.3. Recovery Period
Recovery is a period of time beginning with detection of a lost
packet. Because QUIC retransmits stream data and control frames, not
packets, it defines the end of recovery as a packet sent after the
start of recovery being acknowledged. This is slightly different
from TCP's definition of recovery ending when the lost packet that
started recovery is acknowledged.
During recovery, the congestion window is not increased or decreased.
As such, multiple lost packets only decrease the congestion window
once as long as they're lost before exiting recovery. This causes
QUIC to decrease the congestion window multiple times if
retransmisions are lost, but limits the reduction to once per round
trip.
4.4. Tail Loss Probe
If recovery sends a tail loss probe, no change is made to the
congestion window. Acknowledgement or loss of tail loss probes are
treated like any other packet.
4.5. Retransmission Timeout
When retransmissions are sent due to a retransmission timeout alarm,
no change is made to the congestion window until the next
acknowledgement arrives. The retransmission timeout is considered
spurious when this acknowledgement acknowledges packets sent prior to
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the first retransmission timeout. The retransmission timeout is
considered valid when this acknowledgement acknowledges no packets
sent prior to the first retransmission timeout. In this case, the
congestion window MUST be reduced to the minimum congestion window
and slow start is re-entered.
4.6. Pacing
It is RECOMMENDED that a sender pace sending of all data,
distributing the congestion window over the SRTT. This document does
not specify a pacer. As an example pacer, implementers are referred
to the Fair Queue packet scheduler (fq qdisc) in Linux (3.11 onwards)
as a well-known and publicly available implementation of a flow
pacer.
4.7. Pseudocode4.7.1. Constants of interest
Constants used in congestion control are based on a combination of
RFCs, papers, and common practice. Some may need to be changed or
negotiated in order to better suit a variety of environments.
kDefaultMss (default 1460 bytes): The default max packet size used
for calculating default and minimum congestion windows.
kInitialWindow (default 10 * kDefaultMss): Default limit on the
amount of outstanding data in bytes.
kMinimumWindow (default 2 * kDefaultMss): Default minimum congestion
window.
kLossReductionFactor (default 0.5): Reduction in congestion window
when a new loss event is detected.
4.7.2. Variables of interest
Variables required to implement the congestion control mechanisms are
described in this section.
bytes_in_flight: The sum of the size in bytes of all sent packets
that contain at least one retransmittable or PADDING frame, and
have not been acked or declared lost. The size does not include
IP or UDP overhead. Packets only containing ACK frames do not
count towards byte_in_flight to ensure congestion control does not
impede congestion feedback.
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congestion_window: Maximum number of bytes in flight that may be
sent.
end_of_recovery: The largest packet number sent when QUIC detects a
loss. When a larger packet is acknowledged, QUIC exits recovery.
ssthresh: Slow start threshold in bytes. When the congestion window
is below ssthresh, the mode is slow start and the window grows by
the number of bytes acknowledged.
4.7.3. Initialization
At the beginning of the connection, initialize the congestion control
variables as follows:
congestion_window = kInitialWindow
bytes_in_flight = 0
end_of_recovery = 0
ssthresh = infinite
4.7.4. On Packet Sent
Whenever a packet is sent, and it contains non-ACK frames, the packet
increases bytes_in_flight.
OnPacketSentCC(bytes_sent):
bytes_in_flight += bytes_sent
4.7.5. On Packet Acknowledgement
Invoked from loss detection's OnPacketAcked and is supplied with
acked_packet from sent_packets.
OnPacketAckedCC(acked_packet):
// Remove from bytes_in_flight.
bytes_in_flight -= acked_packet.bytes
if (acked_packet.packet_number < end_of_recovery):
// Do not increase congestion window in recovery period.
return
if (congestion_window < ssthresh):
// Slow start.
congestion_window += acked_packet.bytes
else:
// Congestion avoidance.
congestion_window +=
kDefaultMss * acked_packet.bytes / congestion_window
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